I was wondering if you can pick up a ringing channel by dialing *8# when
you and the other phones are in pickupgroup. Could you do something to
the effect of If the caller was put on a certain extension and just
sitting there... Could you grab the caller by doing something like
*8exten where the
Group
Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going
crazy
Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going
crazy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of PBX
Sent: Friday, November 21, 2003 6:33 PM
To: [EMAIL
on the line and MOH will start.
On Sat, 2003-11-22 at 12:59, PBX wrote:
Is there a solution to have the hold button to play MOH. Or even some
type of ADSI function that allows for this?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy
Aastra 350
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Posted At: Saturday, November 22, 2003 2:08 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going
crazy
Subject: RE:
Hope this helps.
BTW I have an annoying ADSI issue looking for help
with
on Hearing the CAS tone in the voicemail prompts. Do
you hear a loud short Chirp in the vmail prompts on
ADSI phones?
--- PBX [EMAIL PROTECTED] wrote:
Is there any way to program a soft key in ADSI to
put a caller on hold
Ok... I know I have asked this question before, but have never gotten an
answer... When I press the hold button on my phone, should the caller
hear music just like when I park the caller or transfer them to another
extension?
Please assist...
-gcc
___
I know this topic has been covered before - I just want to make sure I
am understanding this correctly.
| ---
|| |
| T1 |Channel Bank |
| --| |
Bank Configuration
On Thu, 2003-11-20 at 19:00, PBX wrote:
I know this topic has been covered before - I just want to make sure I
am understanding this correctly.
| ---
|| |
| T1 |Channel
Is there any way to program a soft key in ADSI to put a caller on hold.
Then able to retreive that caller.
Example -
Softkey Hold
Softkey Retreive Call
Softkey End Call
-gcc
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Configuration
On Thursday 20 November 2003 20:57, PBX wrote:
Example would be -
zaptel.conf -
span=1,0,0,esf,b8zs
EM=1-24
span-2,1,0,esf,b8zs
fxoks=1-24
No:
span=1,0,0,esf,b8zs
span=2,1,0,esf,b8zs
em=1-24
fxoks=25-48
zapata.conf
signalling=fxo_ks
context=blah
group=1
channel = 1-24
Want to put the caller on hold and them hear music by pressing the hold
button
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Posted At: Friday, November 07, 2003 9:56 PM
Posted To: Asterisk User Group
Conversation:
Does anyone know where I can get a list of ADSI functions.. Example *70
(No Call Waiting), Flash = Flash, Hold = ???
Thank you,
-gcc
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If you send out via pine.. Who are you sending the mail out as... Also
if * sends the mail out who is it sending it out as?
Example if you host file only has loopback with localhost then it might
be sent out as [EMAIL PROTECTED] And if Yahoo can resolve
that domain it wont accept the
Pressing hold and the user hears music...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Posted At: Friday, November 07, 2003 9:56 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Putting call on hold
Subject: RE:
Ok.. Example.. I can put them into extension 123 playing MusicOnHold,
but how would I retreive the call when I need to get the caller back?
This is to be done on a analog phone.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Posted
To all Asterisk guru's...
Here is my question.
1. Asterisk PBX - 5 Trunks / Incoming lines
2. 1 Building - 3 Companies (sharing phone system)
Ok that's the basic layout. Here's the low down - Each company will
have one dedicated channel for there company. The other 2 channels they
want
User Group
Conversation: [Asterisk-Users] 5 Channel / Trunk ??
Subject: Re: [Asterisk-Users] 5 Channel / Trunk ??
On Thu, 2003-11-06 at 21:02, PBX wrote:
To all Asterisk guru's...
Here is my question.
1. Asterisk PBX - 5 Trunks / Incoming lines
2. 1 Building - 3 Companies (sharing phone
: [Asterisk-Users] 5 Channel / Trunk ??
Subject: Re: [Asterisk-Users] 5 Channel / Trunk ??
On Thu, 2003-11-06 at 21:02, PBX wrote:
To all Asterisk guru's...
Here is my question.
1. Asterisk PBX - 5 Trunks / Incoming lines
2. 1 Building - 3 Companies (sharing phone system)
Ok that's
Not sure if this is what you are refering to but if so this works for
me
; Tech Support
exten = 8100,1,Dial,Zap/2|15 -- change `15` to what ever
you desire
exten = 8100,2,Voicemail2(u8100)
exten = 8100,102,Voicemail2(b8100)
-gcc
-Original Message-
From: [EMAIL
Update...
Thank you for everyone that replied to this. I received a new 350 today
and plugged it in... Works great. Problem appeared to be the other
phone.
Geoff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Posted At: Thursday, October
/zapata.conf.html
Works the same in http://www.fnords.org/~eric/asterisk/sip.conf.html
On Mon, 2003-10-20 at 23:03, PBX wrote:
I have a quick question...
In the previous thread
http://www.marko.net/asterisk/archives/0210/0306.html it is mentioned
Mark added support for MWI to the chan_zap
I was wondering if anyone has had any experience with PowerTouch 350
ADSI.
I am able to configure the phone with ADSI.. Which is pretty cool in it
self. But I have a question regarding the Service feature. I have to
choose the service button and choose the Asterisk PBX ADSI. But let's
say I
in
depth already
Subject: RE: [Asterisk-Users] MWI - I know this has been discussed in
depth already
If you're using the TDM400P card, VMWI is currently broken...
-Original Message-
From: PBX [mailto:[EMAIL PROTECTED]
Sent: Monday, November 03, 2003 3:17 PM
To: [EMAIL PROTECTED
- PowerTouch 350
What interface is the phone connected to?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of PBX
Sent: Monday, November 03, 2003 8:01 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ADSI - PowerTouch 350
I was wondering
Yes... And I have tried different line cords just rull anything out
Does this make sence why this is doing this.. Could it be the phone it
self is broke?
Geoff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick
Posted At: Thursday, October
I have a question hoping someone can help me with. I just purchased a
350 from http://lktelecom.zoovy.com/product/HPT350. I have been reading
some posts and heard that these phones won't work with Asterisk.. Is
this true? I had a previous Nortel Phone working in this enviroment.
Well anyway
Ok...
The problem is as follows... I received this phone today.
I plugged it in like any other phone that has been used in the past. As
soon as you plug it in and give it power - it starts with a continus
ring - no pause in the ring
(riinn) or
] Nortel PowerTouch 350
Subject: Re: [Asterisk-Users] Nortel PowerTouch 350
On Wed, Oct 29, 2003 at 11:13:38PM -0500, PBX wrote:
Ok...
The problem is as follows... I received this phone today.
I just got mine too... It's an impressive phone.
I plugged it in like any other phone that has been
I have a quick question...
In the previous thread
http://www.marko.net/asterisk/archives/0210/0306.html it is mentioned
Mark added support for MWI to the chan_zap. Is this in the zapata.conf
and if so, if stutter is turned on then the MWI is turned on with it?
Geoff
subscribe to the KISS principle, I do. Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of PBX
Sent: Saturday, October 11, 2003 10:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] T100P Phones Configuration
So...
I would need as you noted two T100P
So...
I would need as you noted two T100P cards or a T400P. The T1 goes into
the * Server and the second port of a T400P goes back to the asterisk
server. Then the extensions get broken out from the Channel bank?
Geoff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I have a question regarding SIP and XTEN softphone.
Server IP - 192.168.1.102
sip.conf --
[7300]
type=friend
host=192.168.1.100
secret=1234
extension.conf --
; Test Sip
exten = 7300,1,Dial,SIP/7300
The sip phones registers, but I am unable to make any calls... Going
either direction... Any
Below you will find, what I believe to be a typical setup with a T100P
card. My question is -
1. Is this correct?
2. What kind of phones would be needed here... (Would you have to use
Digital phones) And if so what would you recommend.
PRI/T1-
|
|
|
The echo canceller algorithms aren't doing anything. We get extreme
echo during the conversation, it appears even before the call connects,
the echo is there...
This only happens with SIP to/from WCFXO (analog POTS). Looking at the
Zaptel configuration:
/etc/asterisk/zapata.conf:
Oh, I forgot to say, zaptel/wcfxo is compiled with:
KFLAGS+=-DECHO_CAN_MARK2
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
(and, Brian, my jack is wired correct..)
-Original Message-
From: Lenny Tropiano [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
PBX
Sent: Sunday, September 21, 2003 5:02 PM
My partner found it!!
Problem solved...
The error was a syntax error in the zapata.conf
channel=1
Should have been written as:
channel=1
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